chromium/third_party/webrtc/modules/audio_coding/neteq/time_stretch.cc

/*
 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#include "modules/audio_coding/neteq/time_stretch.h"

#include <algorithm>  // min, max
#include <memory>

#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "modules/audio_coding/neteq/background_noise.h"
#include "modules/audio_coding/neteq/cross_correlation.h"
#include "modules/audio_coding/neteq/dsp_helper.h"
#include "rtc_base/numerics/safe_conversions.h"

namespace webrtc {

TimeStretch::ReturnCodes TimeStretch::Process(const int16_t* input,
                                              size_t input_len,
                                              bool fast_mode,
                                              AudioMultiVector* output,
                                              size_t* length_change_samples) { … }

void TimeStretch::AutoCorrelation() { … }

bool TimeStretch::SpeechDetection(int32_t vec1_energy,
                                  int32_t vec2_energy,
                                  size_t peak_index,
                                  int scaling) const { … }

}  // namespace webrtc